hi<br>wich codec?<br>once i had a similar problem it was a bandwith problem. <br>how many calls in peak hours? <br>i recomend you tcpdump and then analice the file using wireshark, you will be able to see if the rtp is coming too late or if it inst coming. you can also do the tcpdump in both side so you can see waht is goin out and what is arribing.<br>
<br>if you are transcoding TOO much calls it can be a procesor problem.<br><br>David<br><br><br><br><div class="gmail_quote">2009/2/28 michel freiha <span dir="ltr"><<a href="mailto:michofr@gmail.com">michofr@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir="ltr">Hi all....<br>I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk release all calls...I checked the log file and found..<br>
[Feb 28 11:34:14] NOTICE[19197] chan_sip.c: Disconnecting call 'SIP/netcafe2-b7da99b8' for lack of RTP activity in 301 seconds<br>After that the log show:<br>[Feb 28 11:41:12] WARNING[19197] chan_sip.c: Remote host can't match request CANCEL to call '6697777b27bb46ca01dc42b526adf7bd@Asterisk_IP_Address'. Giving up.<br>
<br>Did someone faced this issue before?<br><br>Thanks for help<br><br>Regards<br></div>
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